Re: [rtcweb] How to multiplex between peers

[Magnus Westerlund <[e-mail]>]

On 2011-07-23 17:12, Stephan Wenger wrote:
> Hi Magnus,
> 
> Not sure you got the context right here.  The "removal of multiparty
> support" was related to my suggestion for a feature list of what I called
> RTPv3, which would need to co-exist with RTPv2 (that includes multiparty
> support).

Ok, I misunderstood that.

> 
> That said, if I'm not completely mistaken, you can implement the
> multiparty use cases using the "RTCP terminating MCU" topology of RFC
> 5117.  This is common implementation practice in video conferencing.  If
> you do so, the multiparty support of RTPv2 is neither exercised nor needed
> in a multiparty video conference.

First of all if you do that you loose functionality as discussed in RFC
5117. For example the indication of who actually sent a packet
originally or contribute to the mix can't be provided unless you let the
SSRC or CSRC plus CNAMES be forwarded across the Mixer/MCU.

In addition it forces the central unit into one particular
implementation, instead of allowing to use what it most appropriate.

An RTP mixer when you want something that has functionality and
application enhancing behavior and the cost is more complexity.

Or you use an relay which simply copies a packet from one entity to all
the others. That is the simplest central node you can build and it will
have no media processing at all.

Also, in any gateway case to existing legacy you don't want to have to
implement additional gatewaying functions just because one want to use
non standard RTP.

For me it is clear that ensuring that RTCWEBs RTP usage is vanilla will
help us avoid a lot of problems down the road.

If we are going to multiplex this mechanism needs to have a behavior
that doesn't change the core behaviors of RTP. If we can do that the
multiplexing becomes much less a hot topic.


Cheers

Magnus Westerlund

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